mirror of
https://github.com/karl0ss/bazarr-ai-sub-generator.git
synced 2025-04-26 14:59:21 +01:00
commit
bf069d1fb4
23
.vscode/launch.json
vendored
23
.vscode/launch.json
vendored
@ -5,16 +5,33 @@
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"version": "0.2.0",
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"configurations": [
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{
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"name": "Python: Current File",
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"name": "Python Debugger: Current File",
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"type": "python",
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"request": "launch",
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"program": "${file}",
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"console": "integratedTerminal",
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"justMyCode": false,
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"env": {
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"CUDA_VISIBLE_DEVICES": "1",
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"LD_LIBRARY_PATH": "/home/karl/faster-auto-subtitle/venv/lib/python3.11/site-packages/nvidia/cublas/lib:/home/karl/faster-auto-subtitle/venv/lib/python3.11/site-packages/nvidia/cudnn/lib"
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},
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"args": [
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"--model",
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"base",
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],
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"base"
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]
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},
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{
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"name": "Current (withenv)",
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"type": "debugpy",
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"request": "launch",
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"program": "${workspaceFolder}/run_with_env.sh",
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"console": "integratedTerminal",
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"justMyCode": false,
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"args": [
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"${file}",
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"--model",
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"base"
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]
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}
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]
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}
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@ -15,16 +15,16 @@ def main():
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parser = argparse.ArgumentParser(
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formatter_class=argparse.ArgumentDefaultsHelpFormatter
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)
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parser.add_argument(
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"--audio_channel", default="0", type=int, help="audio channel index to use"
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)
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parser.add_argument(
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"--sample_interval",
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type=str2timeinterval,
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default=None,
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help="generate subtitles for a specific \
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fragment of the video (e.g. 01:02:05-01:03:45)",
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)
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# parser.add_argument(
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# "--audio_channel", default="0", type=int, help="audio channel index to use"
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# )
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# parser.add_argument(
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# "--sample_interval",
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# type=str2timeinterval,
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# default=None,
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# help="generate subtitles for a specific \
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# fragment of the video (e.g. 01:02:05-01:03:45)",
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# )
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parser.add_argument(
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"--model",
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default="small",
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@ -38,46 +38,27 @@ def main():
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choices=["cpu", "cuda", "auto"],
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help='Device to use for computation ("cpu", "cuda", "auto")',
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)
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# parser.add_argument(
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# "--compute_type",
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# type=str,
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# default="default",
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# choices=[
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# "int8",
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# "int8_float32",
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# "int8_float16",
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# "int8_bfloat16",
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# "int16",
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# "float16",
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# "bfloat16",
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# "float32",
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# ],
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# help="Type to use for computation. \
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# See https://opennmt.net/CTranslate2/quantization.html.",
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# )
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parser.add_argument(
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"--compute_type",
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"--show",
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type=str,
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default="default",
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choices=[
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"int8",
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"int8_float32",
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"int8_float16",
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"int8_bfloat16",
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"int16",
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"float16",
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"bfloat16",
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"float32",
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],
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help="Type to use for computation. \
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See https://opennmt.net/CTranslate2/quantization.html.",
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)
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parser.add_argument(
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"--beam_size",
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type=int,
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default=5,
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help="model parameter, tweak to increase accuracy",
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)
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parser.add_argument(
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"--no_speech_threshold",
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type=float,
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default=0.6,
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help="model parameter, tweak to increase accuracy",
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)
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parser.add_argument(
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"--condition_on_previous_text",
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type=str2bool,
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default=True,
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help="model parameter, tweak to increase accuracy",
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)
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parser.add_argument(
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"--task",
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type=str,
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default="transcribe",
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choices=["transcribe", "translate"],
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default=None,
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help="whether to perform X->X speech recognition ('transcribe') \
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or X->English translation ('translate')",
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)
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@ -6,14 +6,15 @@ from utils.files import filename, write_srt
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from utils.ffmpeg import get_audio, add_subtitles_to_mp4
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from utils.bazarr import get_wanted_episodes, get_episode_details, sync_series
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from utils.sonarr import update_show_in_sonarr
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# from utils.faster_whisper import WhisperAI
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from utils.whisper import WhisperAI
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from utils.decorator import measure_time
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def process(args: dict):
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model_name: str = args.pop("model")
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language: str = args.pop("language")
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sample_interval: str = args.pop("sample_interval")
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audio_channel: str = args.pop("audio_channel")
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show: str = args.pop("show")
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if model_name.endswith(".en"):
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warnings.warn(
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@ -25,26 +26,27 @@ def process(args: dict):
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args["language"] = language
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model_args = {}
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model_args["model_size_or_path"] = model_name
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model_args["device"] = args.pop("device")
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model_args["compute_type"] = args.pop("compute_type")
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list_of_episodes_needing_subtitles = get_wanted_episodes()
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list_of_episodes_needing_subtitles = get_wanted_episodes(show)
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print(
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f"Found {list_of_episodes_needing_subtitles['total']} episodes needing subtitles."
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)
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for episode in list_of_episodes_needing_subtitles["data"]:
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print(f"Processing {episode['seriesTitle']} - {episode['episode_number']}")
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episode_data = get_episode_details(episode["sonarrEpisodeId"])
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audios = get_audio([episode_data["path"]], audio_channel, sample_interval)
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subtitles = get_subtitles(audios, tempfile.gettempdir(), model_args, args)
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add_subtitles_to_mp4(subtitles)
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update_show_in_sonarr(episode["sonarrSeriesId"])
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time.sleep(5)
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sync_series()
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try:
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audios = get_audio([episode_data["path"]], 0, None)
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subtitles = get_subtitles(audios, tempfile.gettempdir(), model_args, args)
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add_subtitles_to_mp4(subtitles)
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update_show_in_sonarr(episode["sonarrSeriesId"])
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time.sleep(5)
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sync_series()
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except Exception as ex:
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print(f"skipping file due to - {ex}")
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@measure_time
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def get_subtitles(
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audio_paths: list, output_dir: str, model_args: dict, transcribe_args: dict
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):
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@ -8,7 +8,7 @@ token = config._sections["bazarr"]["token"]
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base_url = config._sections["bazarr"]["url"]
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def get_wanted_episodes():
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def get_wanted_episodes(show: str=None):
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url = f"{base_url}/api/episodes/wanted"
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payload = {}
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@ -16,7 +16,11 @@ def get_wanted_episodes():
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response = requests.request("GET", url, headers=headers, data=payload)
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return response.json()
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data = response.json()
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if show != None:
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data['data'] = [item for item in data['data'] if item['seriesTitle'] == show]
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data['total'] = len(data['data'])
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return data
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def get_episode_details(episode_id: str):
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13
bazarr-ai-sub-generator/utils/decorator.py
Normal file
13
bazarr-ai-sub-generator/utils/decorator.py
Normal file
@ -0,0 +1,13 @@
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import time
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from datetime import timedelta
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def measure_time(func):
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def wrapper(*args, **kwargs):
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start_time = time.time()
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result = func(*args, **kwargs)
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end_time = time.time()
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duration = end_time - start_time
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human_readable_duration = str(timedelta(seconds=duration))
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print(f"Function '{func.__name__}' executed in: {human_readable_duration}")
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return result
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return wrapper
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68
bazarr-ai-sub-generator/utils/faster_whisper.py
Normal file
68
bazarr-ai-sub-generator/utils/faster_whisper.py
Normal file
@ -0,0 +1,68 @@
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import warnings
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import faster_whisper
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from tqdm import tqdm
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# pylint: disable=R0903
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class WhisperAI:
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"""
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Wrapper class for the Whisper speech recognition model with additional functionality.
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This class provides a high-level interface for transcribing audio files using the Whisper
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speech recognition model. It encapsulates the model instantiation and transcription process,
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allowing users to easily transcribe audio files and iterate over the resulting segments.
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Usage:
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```python
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whisper = WhisperAI(model_args, transcribe_args)
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# Transcribe an audio file and iterate over the segments
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for segment in whisper.transcribe(audio_path):
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# Process each transcription segment
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print(segment)
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```
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Args:
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- model_args: Arguments to pass to WhisperModel initialize method
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- model_size_or_path (str): The name of the Whisper model to use.
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- device (str): The device to use for computation ("cpu", "cuda", "auto").
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- compute_type (str): The type to use for computation.
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See https://opennmt.net/CTranslate2/quantization.html.
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- transcribe_args (dict): Additional arguments to pass to the transcribe method.
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Attributes:
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- model (faster_whisper.WhisperModel): The underlying Whisper speech recognition model.
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- transcribe_args (dict): Additional arguments used for transcribe method.
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Methods:
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- transcribe(audio_path): Transcribes an audio file and yields the resulting segments.
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"""
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def __init__(self, model_args: dict, transcribe_args: dict):
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# self.model = faster_whisper.WhisperModel(**model_args)
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model_size = "base"
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self.model = faster_whisper.WhisperModel(model_size, device="cuda")
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self.transcribe_args = transcribe_args
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def transcribe(self, audio_path: str):
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"""
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Transcribes the specified audio file and yields the resulting segments.
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Args:
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- audio_path (str): The path to the audio file for transcription.
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Yields:
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- faster_whisper.TranscriptionSegment: An individual transcription segment.
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"""
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warnings.filterwarnings("ignore")
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segments, info = self.model.transcribe(audio_path, beam_size=5)
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warnings.filterwarnings("default")
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# Same precision as the Whisper timestamps.
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total_duration = round(info.duration, 2)
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with tqdm(total=total_duration, unit=" seconds") as pbar:
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for segment in segments:
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yield segment
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pbar.update(segment.end - segment.start)
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pbar.update(0)
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@ -7,9 +7,9 @@ def write_srt(transcript: Iterator[dict], file: TextIO):
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for i, segment in enumerate(transcript, start=1):
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print(
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f"{i}\n"
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f"{format_timestamp(segment.start, always_include_hours=True)} --> "
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f"{format_timestamp(segment.end, always_include_hours=True)}\n"
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f"{segment.text.strip().replace('-->', '->')}\n",
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f"{format_timestamp(segment['start'], always_include_hours=True)} --> "
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f"{format_timestamp(segment['end'], always_include_hours=True)}\n"
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f"{segment['text'].strip().replace('-->', '->')}\n",
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file=file,
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flush=True,
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)
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@ -1,9 +1,9 @@
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import warnings
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import faster_whisper
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import torch
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import whisper
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from tqdm import tqdm
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# pylint: disable=R0903
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class WhisperAI:
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"""
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Wrapper class for the Whisper speech recognition model with additional functionality.
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@ -23,23 +23,35 @@ class WhisperAI:
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```
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Args:
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- model_args: Arguments to pass to WhisperModel initialize method
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- model_size_or_path (str): The name of the Whisper model to use.
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- device (str): The device to use for computation ("cpu", "cuda", "auto").
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- compute_type (str): The type to use for computation.
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See https://opennmt.net/CTranslate2/quantization.html.
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- model_args (dict): Arguments to pass to Whisper model initialization
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- model_size (str): The name of the Whisper model to use.
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- device (str): The device to use for computation ("cpu" or "cuda").
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- transcribe_args (dict): Additional arguments to pass to the transcribe method.
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Attributes:
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- model (faster_whisper.WhisperModel): The underlying Whisper speech recognition model.
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- model (whisper.Whisper): The underlying Whisper speech recognition model.
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- device (torch.device): The device to use for computation.
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- transcribe_args (dict): Additional arguments used for transcribe method.
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Methods:
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- transcribe(audio_path): Transcribes an audio file and yields the resulting segments.
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- transcribe(audio_path: str): Transcribes an audio file and yields the resulting segments.
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"""
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def __init__(self, model_args: dict, transcribe_args: dict):
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self.model = faster_whisper.WhisperModel(**model_args)
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"""
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Initializes the WhisperAI instance.
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Args:
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- model_args (dict): Arguments to initialize the Whisper model.
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- transcribe_args (dict): Additional arguments for the transcribe method.
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"""
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device = "cuda" if torch.cuda.is_available() else "cpu"
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print(device)
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# Set device for computation
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self.device = torch.device(device)
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# Load the Whisper model with the specified size
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self.model = whisper.load_model("base.en").to(self.device)
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# Store the additional transcription arguments
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self.transcribe_args = transcribe_args
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def transcribe(self, audio_path: str):
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@ -50,17 +62,24 @@ class WhisperAI:
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- audio_path (str): The path to the audio file for transcription.
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Yields:
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- faster_whisper.TranscriptionSegment: An individual transcription segment.
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- dict: An individual transcription segment.
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"""
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# Suppress warnings during transcription
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warnings.filterwarnings("ignore")
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segments, info = self.model.transcribe(audio_path, **self.transcribe_args)
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# Load and transcribe the audio file
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result = self.model.transcribe(audio_path, **self.transcribe_args)
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# Restore default warning behavior
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warnings.filterwarnings("default")
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# Same precision as the Whisper timestamps.
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total_duration = round(info.duration, 2)
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# Calculate the total duration from the segments
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total_duration = max(segment["end"] for segment in result["segments"])
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# Create a progress bar with the total duration of the audio file
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with tqdm(total=total_duration, unit=" seconds") as pbar:
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for segment in segments:
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for segment in result["segments"]:
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# Yield each transcription segment
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yield segment
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pbar.update(segment.end - segment.start)
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# Update the progress bar with the duration of the current segment
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pbar.update(segment["end"] - segment["start"])
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# Ensure the progress bar reaches 100% upon completion
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pbar.update(0)
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|
@ -1,3 +1,9 @@
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faster-whisper==0.10.0
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tqdm==4.56.0
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ffmpeg-python==0.2.0
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git+https://github.com/openai/whisper.git
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faster-whisper
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nvidia-cublas-cu12
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nvidia-cudnn-cu12
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nvidia-cublas-cu11
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nvidia-cudnn-cu11
|
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ctranslate2==3.24.0
|
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